Hi All.
I've got ser-0.8.99-dev12 working with sems and I have a quick question regarding BYE messages.
How can I send BYE messages"to sems only for SIP transactions that are sems related?
For example, if I dial my own extension and get handed off to seos fmp voicemail, when I hang up
the phone, I need to make sure a BYE get sent to sems so that it can to clean up work if needef
when tearing down the call.
So my problem is that I do not know how to differentiate BYE messages from sems related calls
versus non-sems related calls.
I was thinking about append_hf("P-hint: SEMS-CALL") or something, but these headers don't seem to
get propagated from the original INVITE to the BYE.
Regards,
Paul
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I believe that three places in /serweb_2004-07-27/html/config.php
the string "/user" should be "user_interface/". The important one is:
/* relative paths of serweb tree */
$this->img_src_path = $this->root_path."img/";
$this->js_src_path = $this->root_path."styles/";
$this->style_src_path = $this->root_path."styles/";
$this->user_pages_path = $this->root_path."user/";
-----------------------------------------------------------^^^^^ HERE
$this->admin_pages_path = $this->root_path."admin/";
It causes the admin pages to have bad links.
John.
Hi Jiang,
very helpful will be complete log of ser - use debug=9 to get more info.
Somewhere in the log should be some references to the error.
Best regards,
Marian
Jiang zhou wrote:
> Hi, Marian
>
> I use xten's x-lite as client. I configure is with authen user:test and
> passwod:test. But it register failed. I have dump the
> packet ser sent and recived.
>
--
Voice Sistem
http://www.voice-sistem.ro
At least on my server the error messages about lookup("aliases") and
save("aliases") occur on the first time a user connects to confirm
subscription, eg by clicking on the link sent to them by email. If the
user clicks a second time, the error goes away. Anyone else seen
this? Any idea how to fix it? (My ser.cfg is very close to the default
setup with mysql enabled).
John.
I switched from 0.8.12 to 0.8.14, no issues.. working fine (no
improvement though :-)) but that probably because I did not use
something weird/complicated :-)
________________________________
From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On
Behalf Of Darren Nay
Sent: Wednesday, November 10, 2004 4:14 PM
To: serusers(a)lists.iptel.org
Subject: [Serusers] Upgrade to 0.8.14
Hey all -
Are there any compatibility issues that I should be aware of when
upgrading from 0.8.12 to 0.8.14?
We are working very well on 0.8.12 currently. I just want to evaluate
upgrading. Are there bug fixes in 0.8.14 that are not in 0.8.12?
Basically, what will I gain from upgrading, and what problems will I run
into?
Thanks,
Darren Nay
VOIP Network Development
Ionosphere, Inc.
Rafael,
As i said in my previous mail, it just was a guess. Can you try modifying your route[1] something
like this. I think this will send only the unanswered calls to voicemail. All other 4** status
will be ignored. I havent tested it though.
Also, you were right. It is route[1], not route[4].
route[1] {
if (t_check_status("408")){
revert_uri();
rewritehostport("200.110.2.132:5060");
append_branch();
t_relay ();
break;
}
}
Good Luck,
--- "Rafael J. Risco G.V." <rafael.risco(a)gmail.com> wrote:
> hello Girish
> I did some test using t_check_status ("487") in failure_route[1] like this:
>
> failure_route[1] {
> if (t_check_status("487")){
> revert_uri ();
> rewritehostport ("200.110.2.132:5060");
> append_branch();
> t_relay_to_udp("200.110.2.132", "5060");
> }
>
> ...but did not work, can you send us more details about how to use
> t_check_status("487") in route[4]?
>
> thanks
> rafael
>
>
> On Tue, 9 Nov 2004 02:25:26 -0800 (PST), Girish <gr_sh2003(a)yahoo.com> wrote:
> > Hello,
> >
> > Just a guess.. Can you add t_check_status (487) in your route[4] and see if the INVITE goes to
> > Asterisk after cancelling?
> >
> >
> >
> > --- "Rafael J. Risco G.V." <rafael.risco(a)gmail.com> wrote:
> >
> > > Hello
> > > I have to insist in this issue since I�ve done several test using Sems
> > > and asterisk with very simple configuration files including the
> > > original example from ser-cvs... in brief: if I call to a user who
> > > belongs to voicemail group and I cancel the call before VM forward
> > > routine begin then an "invite" is sent to a voicemail server
> > > generating and sending a file with No audio, and I cant account this
> > > call with "Sip-Response-Code=487" (just an start record without
> > > stop)...
> > >
> > > does someone know how to solve this problem????
> > >
> > > thanks in advance
> > >
> > > Rafael
> > >
> > > PS:
> > > ser.cfg and asterisk debug for this test:
> > >
> > > #
> > > # SER SIMPLE CFG for VM without acc...
> > > # ----------- global configuration parameters ------------------------
> > >
> > > #debug=3 # debug level (cmd line: -dddddddddd)
> > > #fork=yes
> > > #log_stderror=no # (cmd line: -E)
> > >
> > > #/* Uncomment these lines to enter debugging mode
> > > debug=9
> > > fork=yes
> > > log_stderror=yes
> > > #*/
> > >
> > > listen=127.0.0.1
> > > port=5060
> > >
> > > # simple proxy script for forwarding to voicemail server
> > > # for unavailable users
> > > #
> > >
> > > loadmodule "/usr/local/lib/ser/modules/sl.so"
> > > loadmodule "/usr/local/lib/ser/modules/tm.so"
> > > loadmodule "/usr/local/lib/ser/modules/rr.so"
> > > loadmodule "/usr/local/lib/ser/modules/maxfwd.so"
> > > loadmodule "/usr/local/lib/ser/modules/mysql.so"
> > > loadmodule "/usr/local/lib/ser/modules/group.so"
> > > loadmodule "/usr/local/lib/ser/modules/usrloc.so"
> > > loadmodule "/usr/local/lib/ser/modules/registrar.so"
> > >
> > > # time to give up on ringing -- global timer, applies to
> > > # all transactions
> > > modparam("tm", "fr_inv_timer", 90)
> > >
> > > # database with user group membership
> > > modparam("group", "db_url", "mysql://ser:heslo@localhost/ser")
> > >
> > >
> > > # --------------------- request routing logic -------------------
> > > route {
> > >
> > > if (!mf_process_maxfwd_header("10")) {
> > > log("LOG: Too many hops\n");
> > > sl_send_reply("483", "Alas Too Many Hops");
> > > break;
> > > };
> > >
> > > if (!(method=="REGISTER")) record_route();
> > > if (loose_route()) {
> > > t_relay();
> > > break;
> > > };
> > >
> > > if (!uri==myself) {
> > > t_relay();
> > > break;
> > > };
> > >
> > > if (method == "REGISTER") {
> > > if (!save("location")) {
> > > sl_reply_error();
> > > };
> > > break;
> > > };
> > >
> > > # does the user wish redirection on no availability? (i.e., is he
> > > # in the voicemail group?) -- determine it now and store it in
> > > # flag 4, before we rewrite the flag using UsrLoc
> > > if (is_user_in("Request-URI", "voicemail")) {
> > > setflag(4);
> > > };
> > >
> > > # native SIP destinations are handled using our USRLOC DB
> > > if (!lookup("location")) {
> > > # handle user which was not found
> > > route(4);
> > > break;
> > > };
> > >
> > > # if user is on-line and is in voicemail group, enable redirection
> > > if (method == "INVITE" && isflagset(4)) {
> > > t_on_failure("1");
> > > };
> > > t_relay();
> > > }
> > >
> > > # ------------- handling of unavailable user ------------------
> > > route[4] {
> > >
> > > # non-Voip -- just send "off-line"
> > > if (!(method == "INVITE" || method == "ACK" || method == "CANCEL")) {
> > > sl_send_reply("404", "Not Found");
> > > break;
> > > };
> > >
> > > # not voicemail subscriber
> > > if (!isflagset(4)) {
> > > sl_send_reply("404", "Not Found and no voicemail turned on");
> > > break;
> > > };
> > >
> > > # forward to voicemail now
> > > rewritehostport("200.110.2.132:5060");
> > > t_relay_to_udp("200.110.2.132", "5060");
> > > }
> > >
> > > # if forwarding downstream did not succeed, try voicemail running
> > > # at 200.110.2.132:5060
> > >
> > > failure_route[1] {
> > > revert_uri();
> > > rewritehostport("200.110.2.132:5060");
> > > append_branch();
> > > t_relay_to_udp("200.110.2.132", "5060");
> > > }
> > >
> > >
=====
Girish Gopinath <gr_sh2003(a)yahoo.com>
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Hello,
I must do a monitoring tool for a SER sip proxy. I have a user and a pass to
it. I can make calls with the combination.
Tests I-ve ran:
rhemaxos root # sipsak -n -s sip:me@myhost:5060 -v
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.0.1:1675;rport=
From: <sip:sipsak@127.0.0.1:1675>;tag=2fbba8ac
To: <sip:**************:5060>
CSeq: 1 OPTIONS
Call-ID: 800827564(a)127.0.0.1
Content-Length: 0
All seem ok
but if i run:
rhemaxos root # sipsak -I -U -n -s sip:me@myhost:5060 -v -a mypass
i get smth like:
Call-ID: 2105779733(a)127.0.0.1
WWW-Authenticate: DIGEST realm="whatever.com",
nonce="fde24b2e80f3f1c96ec8e962d34627d0", algorithm=MD5, opaque="3251298552"
Content-Length: 0
error: couldn't find authentication method Digest in the 402 response above
The dumps (i use nast) from Xlite look a lot different.
Well, really beats me, i'm new to voip, and i really need help with this
one. It does not even try to AUTH me.
I know that is my fault, but i need some help with this one. I would not
like to read the SIP RFC just to monitor a SIP proxy. I have basic to medium
theoretical knowledge of the protocols and no experience in the wild.
Thank you in advance
Dragos
Hey all -
Are there any compatibility issues that I should be aware of when upgrading
from 0.8.12 to 0.8.14?
We are working very well on 0.8.12 currently. I just want to evaluate
upgrading. Are there bug fixes in 0.8.14 that are not in 0.8.12?
Basically, what will I gain from upgrading, and what problems will I run
into?
Thanks,
Darren Nay
VOIP Network Development
Ionosphere, Inc.
Hello,
I have attached my modified natehper.c file.
I have tested with 2 Eyepmedia soft phones.
After your test, can you say me if it is ok and with which
phones/softPhones.
Thanks
Laurent
-----Original Message-----
From: Zhou Jun [mailto:ottife@vip.sina.com]
Sent: mercredi, 10. novembre 2004 16:21
To: laurent.schweizer(a)eivd.ch
Subject: RE: [Serusers] rtpproxy+nathelper (0.8.14) +video
Hello Laurent,
I wait for this patch a long time.
Could you share your work to me?
Thanx much.
BR//Juny
----- Original Message -----
> Date: Mon, 8 Nov 2004 17:18:41 +0100
> From: "Schweizer Laurent" <laurent.schweizer(a)eivd.ch>
> Subject: RE: [Serusers] rtpproxy+nathelper (0.8.14) +video
> To: "'Zeus Ng'" <zeus.ng(a)isquare.com.au>
> Cc: serusers(a)lists.iptel.org
> Message-ID: <EINT11A81Np7vXee4fD00000443(a)EINT11.einet.ad.eivd.ch>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hi, I made some changes to the nathelper and now the video is working, I
> have tested this with the Eyepmedia softphone.
>
> if somebody is interested...
>
> Laurent
